diff options
| -rw-r--r-- | .gitignore | 1 | ||||
| -rw-r--r-- | Cargo.lock | 468 | ||||
| -rw-r--r-- | Cargo.toml | 13 | ||||
| -rw-r--r-- | src/decoder.rs | 442 | ||||
| -rw-r--r-- | src/main.rs | 72 |
5 files changed, 996 insertions, 0 deletions
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filter, STFT for pitch detection, +// a Goertzel filter for tone extraction, and the core decoding logic. + +use anyhow::{Result, bail}; +use collections::VecDeque; +use rubato::{InterpolationParameters, InterpolationType, Resampler, SincFixedIn, WindowFunction}; +use rustfft::{Fft, FftPlanner, num_complex::Complex}; +use std::sync::Arc; + +// --- DSP Constants (ported from ggmorse) --- +const FREQ_MIN_HZ: f32 = 200.0; +const FREQ_MAX_HZ: f32 = 1200.0; +const HISTORY_S: f32 = 3.0; // How many seconds of audio to keep for analysis + +// --- Biquad Filter (Corrected Implementation) --- +#[derive(Debug, Clone, Copy)] +pub enum FilterType { + HighPass, + LowPass, +} + +pub struct BiquadFilter { + a0: f32, + a1: f32, + a2: f32, + b1: f32, + b2: f32, + x1: f32, + x2: f32, // Delayed inputs + y1: f32, + y2: f32, // Delayed outputs +} + +impl BiquadFilter { + pub fn new(filter_type: FilterType, cutoff_hz: f32, sample_rate: u32) -> Self { + let mut filter = Self { + a0: 1.0, + a1: 0.0, + a2: 0.0, + b1: 0.0, + b2: 0.0, + x1: 0.0, + x2: 0.0, + y1: 0.0, + y2: 0.0, + }; + let c = (std::f32::consts::PI * cutoff_hz / sample_rate as f32).tan(); + let sqrt2 = 2.0f32.sqrt(); + + match filter_type { + FilterType::LowPass => { + let d = 1.0 / (1.0 + sqrt2 * c + c * c); + filter.a0 = c * c * d; + filter.a1 = 2.0 * filter.a0; + filter.a2 = filter.a0; + filter.b1 = 2.0 * (c * c - 1.0) * d; + filter.b2 = (1.0 - sqrt2 * c + c * c) * d; + } + FilterType::HighPass => { + let d = 1.0 / (1.0 + sqrt2 * c + c * c); + filter.a0 = d; + filter.a1 = -2.0 * d; + filter.a2 = d; + filter.b1 = 2.0 * (c * c - 1.0) * d; + filter.b2 = (1.0 - sqrt2 * c + c * c) * d; + } + } + filter + } + + pub fn process(&mut self, input: &mut [f32]) { + for sample in input.iter_mut() { + let x0 = *sample; + let y0 = self.a0 * x0 + self.a1 * self.x1 + self.a2 * self.x2 + - self.b1 * self.y1 + - self.b2 * self.y2; + + self.x2 = self.x1; + self.x1 = x0; + self.y2 = self.y1; + self.y1 = y0; + + *sample = y0; + } + } +} + +// --- Goertzel Filter --- +struct Goertzel { + coeff: f32, + history: VecDeque<f32>, + window: Vec<f32>, +} + +impl Goertzel { + fn new(target_freq: f32, sample_rate: u32, window_size: usize) -> Self { + let k = (0.5 + (window_size as f32 * target_freq) / sample_rate as f32) as usize; + let omega = (2.0 * std::f32::consts::PI * k as f32) / window_size as f32; + let coeff = 2.0 * omega.cos(); + + let window = (0..window_size) + .map(|i| { + 0.54 - 0.46 * (2.0 * std::f32::consts::PI * i as f32 / window_size as f32).cos() + }) + .collect(); + + Self { + coeff, + history: VecDeque::with_capacity(window_size), + window, + } + } + + fn run(&self, samples: &[f32]) -> f32 { + let mut q0; + let mut q1 = 0.0; + let mut q2 = 0.0; + + for (i, &sample) in samples.iter().enumerate() { + q0 = self.coeff * q1 - q2 + sample * self.window[i]; + q2 = q1; + q1 = q0; + } + + q1 * q1 + q2 * q2 - self.coeff * q1 * q2 + } + + fn process_stream(&mut self, samples: &[f32]) -> Vec<f32> { + self.history.extend(samples.iter()); + let mut power = Vec::new(); + while self.history.len() >= self.window.len() { + let chunk: Vec<f32> = self + .history + .iter() + .take(self.window.len()) + .copied() + .collect(); + power.push(self.run(&chunk)); + self.history.pop_front(); + } + power + } +} + +// --- Main Decoder --- +pub struct MorseDecoder { + resampler: SincFixedIn<f32>, + filter_hp: BiquadFilter, + filter_lp: BiquadFilter, + audio_buffer: Vec<f32>, + target_sample_rate: u32, + estimated_pitch: Option<f32>, +} + +impl MorseDecoder { + pub fn new(source_sample_rate: u32, target_sample_rate: u32) -> Result<Self> { + let resampler = if source_sample_rate != target_sample_rate { + let params = InterpolationParameters { + sinc_len: 256, + f_cutoff: 0.95, + interpolation: InterpolationType::Linear, + oversampling_factor: 256, + window: WindowFunction::BlackmanHarris, + }; + SincFixedIn::new( + target_sample_rate as f64 / source_sample_rate as f64, + 2.0, + params, + 1024, // chunk size + 1, // channels + )? + } else { + // Create a dummy resampler if not needed + SincFixedIn::new( + 1.0, + 1.0, + InterpolationParameters { + sinc_len: 2, + ..Default::default() + }, + 1024, + 1, + )? + }; + + Ok(Self { + resampler, + filter_hp: BiquadFilter::new(FilterType::HighPass, FREQ_MIN_HZ, target_sample_rate), + filter_lp: BiquadFilter::new(FilterType::LowPass, FREQ_MAX_HZ, target_sample_rate), + audio_buffer: Vec::new(), + target_sample_rate, + estimated_pitch: None, + }) + } + + pub fn process(&mut self, chunk: &[f32]) -> Result<()> { + let waves_in = vec![chunk.to_vec()]; + let mut resampled = self.resampler.process(&waves_in, None)?; + + let mut audio_chunk = resampled.remove(0); + self.filter_hp.process(&mut audio_chunk); + self.filter_lp.process(&mut audio_chunk); + + self.audio_buffer.extend(audio_chunk); + Ok(()) + } + + pub fn finalize(&mut self) -> Result<String> { + if self.audio_buffer.is_empty() { + return Ok(String::new()); + } + + // 1. Detect Pitch using STFT + let pitch = self.detect_pitch_stft()?; + log::info!("Estimated pitch: {:.2} Hz", pitch); + self.estimated_pitch = Some(pitch); + + // 2. Extract signal power using Goertzel filter + let goertzel_window_size = (self.target_sample_rate / 50) as usize; // ~20ms window + let mut goertzel_filter = + Goertzel::new(pitch, self.target_sample_rate, goertzel_window_size); + let power_signal = goertzel_filter.process_stream(&self.audio_buffer); + + // 3. Find optimal WPM and Threshold + let (best_wpm, best_threshold) = self.find_best_params(&power_signal)?; + log::info!( + "Best fit: WPM = {}, Threshold = {:.4}", + best_wpm, + best_threshold + ); + + // 4. Decode with optimal parameters + let text = self.decode_with_params(&power_signal, best_wpm, best_threshold); + + Ok(text) + } + + fn detect_pitch_stft(&self) -> Result<f32> { + let fft_size = 2048; + let step_size = fft_size / 4; + + let mut planner = FftPlanner::new(); + let fft = planner.plan_fft_forward(fft_size); + let window: Vec<f32> = (0..fft_size) + .map(|i| 0.54 - 0.46 * (2.0 * std::f32::consts::PI * i as f32 / fft_size as f32).cos()) + .collect(); + + let mut spectrum_sum = vec![0.0; fft_size / 2]; + let mut count = 0; + + for chunk in self.audio_buffer.windows(fft_size).step_by(step_size) { + let mut buffer: Vec<Complex<f32>> = chunk + .iter() + .zip(window.iter()) + .map(|(s, w)| Complex::new(s * w, 0.0)) + .collect(); + + fft.process(&mut buffer); + + for (i, v) in buffer.iter().take(fft_size / 2).enumerate() { + spectrum_sum[i] += v.norm_sqr(); + } + count += 1; + } + + if count == 0 { + bail!("Not enough audio data to detect pitch."); + } + + let df = self.target_sample_rate as f32 / fft_size as f32; + let mut max_power = 0.0; + let mut best_freq = 0.0; + + for i in 0..fft_size / 2 { + let freq = i as f32 * df; + if freq >= FREQ_MIN_HZ && freq <= FREQ_MAX_HZ { + let power = spectrum_sum[i]; + if power > max_power { + max_power = power; + best_freq = freq; + } + } + } + + Ok(best_freq) + } + + fn find_best_params(&self, power_signal: &[f32]) -> Result<(f32, f32)> { + let mut best_cost = f32::MAX; + let mut best_wpm = 20.0; + let mut best_threshold = 0.0; + + let mean_power = power_signal.iter().sum::<f32>() / power_signal.len() as f32; + + for wpm_int in 5..=40 { + let wpm = wpm_int as f32; + for l in (10..=90).step_by(5) { + let threshold = mean_power * (l as f32 / 100.0); + let cost = self.calculate_cost(power_signal, wpm, threshold); + if cost < best_cost { + best_cost = cost; + best_wpm = wpm; + best_threshold = threshold; + } + } + } + Ok((best_wpm, best_threshold)) + } + + fn calculate_cost(&self, power_signal: &[f32], wpm: f32, threshold: f32) -> f32 { + let dot_len_ms = 1200.0 / wpm; + let dot_len_samples = (dot_len_ms / 1000.0) + * (self.target_sample_rate as f32 / ((self.target_sample_rate / 50) as f32)); + + let mut on_intervals = Vec::new(); + let mut off_intervals = Vec::new(); + let mut current_len = 0; + let mut is_on = power_signal[0] > threshold; + + for &p in power_signal { + if (p > threshold) == is_on { + current_len += 1; + } else { + if is_on { + on_intervals.push(current_len); + } else { + off_intervals.push(current_len); + } + is_on = !is_on; + current_len = 1; + } + } + + if on_intervals.is_empty() { + return f32::MAX; + } + + let cost_on: f32 = on_intervals + .iter() + .map(|&len| { + let cost_dot = (len as f32 / dot_len_samples - 1.0).powi(2); + let cost_dash = (len as f32 / dot_len_samples - 3.0).powi(2); + cost_dot.min(cost_dash) + }) + .sum(); + + cost_on / on_intervals.len() as f32 + } + + fn decode_with_params(&self, power_signal: &[f32], wpm: f32, threshold: f32) -> String { + let dot_len_ms = 1200.0 / wpm; + // The power signal has a lower sample rate because of the Goertzel windowing + let power_signal_rate = + self.target_sample_rate as f32 / (self.target_sample_rate as f32 / 50.0); + let dot_len_samples = (dot_len_ms / 1000.0) * power_signal_rate; + + let mut result = String::new(); + let mut current_letter = String::new(); + + let mut current_len = 0; + let mut is_on = power_signal[0] > threshold; + + for &p in power_signal.iter().chain(std::iter::once(&0.0)) { + // Add sentinel + if (p > threshold) == is_on { + current_len += 1; + } else { + let len_norm = current_len as f32 / dot_len_samples; + if is_on { + // end of a tone + if (len_norm - 1.0).abs() < (len_norm - 3.0).abs() { + current_letter.push('0'); // dot + } else { + current_letter.push('1'); // dash + } + } else { + // end of a space + if len_norm > 2.0 { + // inter-letter space + if let Some(c) = morse_to_char(¤t_letter) { + result.push(c); + } else if !current_letter.is_empty() { + result.push('?'); // Unknown character + } + current_letter.clear(); + if len_norm > 5.0 { + // word space + result.push(' '); + } + } + // else, it's an inter-element space, do nothing + } + is_on = !is_on; + current_len = 1; + } + } + result + } +} + +fn morse_to_char(s: &str) -> Option<char> { + match s { + "01" => Some('A'), + "1000" => Some('B'), + "1010" => Some('C'), + "100" => Some('D'), + "0" => Some('E'), + "0010" => Some('F'), + "110" => Some('G'), + "0000" => Some('H'), + "00" => Some('I'), + "0111" => Some('J'), + "101" => Some('K'), + "0100" => Some('L'), + "11" => Some('M'), + "10" => Some('N'), + "111" => Some('O'), + "0110" => Some('P'), + "1101" => Some('Q'), + "010" => Some('R'), + "000" => Some('S'), + "1" => Some('T'), + "001" => Some('U'), + "0001" => Some('V'), + "011" => Some('W'), + "1001" => Some('X'), + "1011" => Some('Y'), + "1100" => Some('Z'), + "01111" => Some('1'), + "00111" => Some('2'), + "00011" => Some('3'), + "00001" => Some('4'), + "00000" => Some('5'), + "10000" => Some('6'), + "11000" => Some('7'), + "11100" => Some('8'), + "11110" => Some('9'), + "11111" => Some('0'), + _ => None, + } +} diff --git a/src/main.rs b/src/main.rs new file mode 100644 index 0000000..d288331 --- /dev/null +++ b/src/main.rs @@ -0,0 +1,72 @@ +use anyhow::{bail, Result}; +use clap::Parser; +use hound::{SampleFormat, WavReader}; +use std::path::PathBuf; + +mod decoder; +use decoder::MorseDecoder; + +const TARGET_SAMPLE_RATE: u32 = 12000; // Same as ggmorse's kBaseSampleRate +const CHUNK_SIZE: usize = 4096; + +#[derive(Parser)] +#[command(author, version, about, long_about = None)] +struct Cli { + /// Path to the input WAV file + #[arg(value_name = "WAV_FILE")] + wav_file: PathBuf, +} + +fn main() -> Result<()> { + // Set up logging. Use `RUST_LOG=info` or `RUST_LOG=debug` to see output. + env_logger::init(); + let cli = Cli::parse(); + + log::info!("Opening WAV file: {:?}", cli.wav_file); + let mut reader = WavReader::open(cli.wav_file)?; + let spec = reader.spec(); + log::info!("WAV spec: {:?}", spec); + + if spec.sample_format != SampleFormat::Int && spec.sample_format != SampleFormat::Float { + bail!( + "Unsupported sample format: {:?}. Only 16-bit Int and 32-bit Float are supported.", + spec.sample_format + ); + } + + // --- Create the Morse Decoder --- + let mut decoder = MorseDecoder::new(spec.sample_rate, TARGET_SAMPLE_RATE)?; + + // --- Read and process the audio in chunks --- + let samples_f32: Vec<f32> = if spec.sample_format == SampleFormat::Int { + reader + .samples::<i16>() + .map(|s| s.unwrap() as f32 / 32768.0) + .collect() + } else { + reader.samples::<f32>().map(|s| s.unwrap()).collect() + }; + + // Convert to mono by averaging channels if necessary + let mono_samples: Vec<f32> = if spec.channels > 1 { + samples_f32 + .chunks_exact(spec.channels as usize) + .map(|chunk| chunk.iter().sum::<f32>() / spec.channels as f32) + .collect() + } else { + samples_f32 + }; + + // Process the entire audio buffer + for chunk in mono_samples.chunks(CHUNK_SIZE) { + decoder.process(chunk)?; + } + + // Finalize decoding after all audio is processed + let decoded_text = decoder.finalize()?; + + println!("\n--- Decoded Text ---"); + println!("{}", decoded_text); + + Ok(()) +}
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